Re: [rtcweb] Proposal to break the ICE impasse

Ted Hardie <ted.ietf@gmail.com> Thu, 24 January 2019 20:59 UTC

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From: Ted Hardie <ted.ietf@gmail.com>
Date: Thu, 24 Jan 2019 12:58:38 -0800
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To: Adam Roach <adam@nostrum.com>
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Subject: Re: [rtcweb] Proposal to break the ICE impasse
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For what it is worth, I think this way forward should work reasonably
well.  It doesn't create a conflict for the current implementations, but
does leave the breadcrumbs needed so that any implementations in the future
that do add support for this will do so interoperably.

If folks can review the PR (it's quite short) and comment, I would
appreciate it.  Assuming we get consensus, I think a quick re-spin is all
that's necessary to move forward.

thanks,

Ted

On Thu, Jan 24, 2019 at 11:12 AM Adam Roach <adam@nostrum.com> wrote:

> Based on conversations in MMUSIC, as well as several offline
> conversations with interested parties, I've put together a proposed
> change to JSEP that, if accepted, will allow publication of the Cluster
> 238 documents to move forward.
>
> Note that this new text has no impact on existing implementations (at
> least, as far as I am able to discern), which do not currently have the
> capability of producing media sections consisting of exclusively TCP
> candidates. From that perspective, the change makes existing
> implementations no less compliant with JSEP than they were before.
>
> What this change does provide is both on-paper and in-the-future
> compatibility between WebRTC implementations once they finalize TCP
> candidate handling (and candidate handling in general for mid-session
> offers).
>
>    https://github.com/rtcweb-wg/jsep/pull/862/files
>
> The key insight here is that JSEP's use of ICE completely discards any
> meaning associated with the transport parameter, while SIP's use of ICE
> does not. The trivial change that I propose, which bears only on future
> WebRTC implementations -- that is, which has no as-built specification
> to point to -- allows JSEP to continue to ignore the value of the
> transport parameter, while specifying that it says the right thing for
> SIP implementations to function properly.
>
> /a
>
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> rtcweb@ietf.org
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